Re: [sipx-users] "Process 'SipXbridge' stopped unexpectedly. Attempting to restart the process."

spix is operational. Our server is not acting as DNS server. The missing records seem to be SRV and NAPTAR. Not sure. Thanks Nikolay From:"Nikolay Kondratyev" To:"'Discussion list for users of sipXecs software'" Date:05/04/2011 09:59 A

Re: [sipx-users] Pick-up group?

If the call is going to "extension" not "hunt group", will "*78" pick up the call?. And moreover, what if the call is going to the phone, that is in several hunt groups? Which number should one dial to pick up the call? And I don't want to remember "hunt group number" to pick up a call. There

Re: [sipx-users] "Process 'SipXbridge' stopped unexpectedly. Attempting to restart the process."

Not sure if absence of dns record is related to sipxbridge alarm, but in general sipx will not work without correct dns configuration. Does your sipx serve as dns server? Which DNS records are missing (according to dns advisor)? Rgds, Nikolay. From: sipx-users-boun...@list.sipfoundry.org

Re: [sipx-users] ITSP registration expires before reregistration, calls are dropped

Anders, I tried to look into your trace. Actually I don't see any problems in the trace. Strange is that some register messages are challenged (401) by the itsp, and some are not. Another strange idea is to use mix of public and private ip addresses without nat. I would not do that in any

Re: [sipx-users] IANT click to dial on win 7

7;ve fixed and uploaded a new version. Just uninstall your old version and install the new from our website. Thanks for your feedback J Regards Jan Fricke Von: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] Im Auftrag von Nikolay Kondraty

[sipx-users] {Disarmed} RE: {Disarmed} Reboot phone function - doesn't work for Linksys ATAs

Sipx sends Notify message with check-sync header to the phone. It works only for registered phones. My spa942 phone was by default configured to challenge that Notify, and sipx can not handle that challenge. So there is a possibility, that you just have to configure your spa ata's not to challeng

Re: [sipx-users] Ghost Calls

XX-6698 is marked as fixed in 4.2. You run "latest production" version? What is it? Is it 4.2.1? Why do you think you encountered XX-6698? Did you compare traces? Do you see those calls in sipx cdr? That is, it may happen that your ghost calls go from "something" directly to your pstn gateway.

Re: [sipx-users] SBC or not

The decision is quite up to you. But you may want to take into account the following: Sipxbridge will in certain sense isolate your sipx from the telco. Strictly speaking it will do the following: 1. Terminate sip signaling (Telco will talk to the sipxbridge, not to the phones) 2. Force media rela

Re: [sipx-users] Lookup name for incoming call

Freeswitch is capable of doing that. Since freeswitch is now part of sipx installation, you could route a call to a separate freeswitch instance and do what you need, and then route a call back to sipxproxy. You may find some examples here http://wiki.sipfoundry.org/display/sipXecs/Custom+FreeSWI

Re: [sipx-users] hunt group issue...

It may happen that sipx 4.2.1 challenges calls from mediant. 3.10 did not do that, as far as I remember. If this is the case, you can configure mediant to handle those challenges. Rgds, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoun

Re: [sipx-users] Help with Cisco 7905 / Softphone

Hi, I tried to look into your traces. I did not find anything wrong there. What is there in the network trace? Does 7905 really receive rtp flow? Rgds, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Alistair J.

Re: [sipx-users] Problem with calls via GW when modifying "DefaultCaller ID"

Are they (linksys/cisco) going to release "bug-fix" firmware version for the phone? Rgds, Nikolay. > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Niklas > Sent: Monday, February 14, 2011 4:29 PM > To: sipx

Re: [sipx-users] Audiocodes IP phones

I tried 320HD phones some time ago. The quality of voice was quite good. IMO, in comparison to new Grandstream 21xx HD phones that I test now, AC phones voice is much better. AC phones do not support "phone initiated MoH". I did not test if BLF on AC phones works with sipx. Rgds, Nikolay. >

Re: [sipx-users] attended transfer fails under certain conditions

Could you send a trace (xml) of that failed scenario? Rgds, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Scheidell Sent: Tuesday, February 08, 2011 11:11 PM To: sipx-users@list.sipfoundry.org users Subject: [

Re: [sipx-users] Problem with calls via GW when modifying "DefaultCaller ID"

And with 6.1.5 spa942 still does not send Ack? Could you send a new trace? I heard that spa942 do not like long sip messages (and may be long sip headers). But I don't now the limitation. So... I would guess, that due to "caller id substitution" wery long Record-Route header occurs. And that's why

Re: [sipx-users] Problem with calls via GW when modifying "DefaultCaller ID"

Niklas, I don't see my message I sent to the list yesterday. Resending it... Niklas, you are using old firmware on your spa942. Upgrade it to 6.1.5 (or later, if exists). There is a possibility, that after that spa942 will start to send Ack. :) If not, take a new trace and let's see what's there.

Re: [sipx-users] Problem with calls via GW when modifying "DefaultCaller ID"

Niklas, First, you are using old firmware on your spa942. Upgrade it to 6.1.5 (or later, if exists). There is a possibility, that after that spa942 will start to send Ack. :) If not, take a new trace and let's see what's there... Rgds, Nikolay. > -Original Message- > From: sipx-users-boun

Re: [sipx-users] phone INVITE -> TRYING -> RINGING -> connect? but no audio/voice with PolyCom SoundPoint 650

As far as I understand smartnode 4638 is bri voip gateway. If you plan to connect to PSTN only through this bri gw, and do not plan to use sip-trunks to ITSP's, you may want to disable siptrunking role for the sipx. Anyway read about sipxbridge and siptrunking on the wiki. And about far/near end na

Re: [sipx-users] phone INVITE -> TRYING -> RINGING -> connect?but no audio/voice with PolyCom SoundPoint 650

uch time... ;) ) Rgds, Nikolay. > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of > Nikolay Kondratyev > Sent: Friday, February 04, 2011 3:02 PM > To: 'Discussion list for users of

Re: [sipx-users] phone INVITE -> TRYING -> RINGING -> connect? but no audio/voice with PolyCom SoundPoint 650

Ok, now I see all signalling messages in pcap files. (unfortunately your xml trace file does not contain the whole call...). I do not change my opinion: You do not have voice because one of your polycom phones (200) does not send Ack message. Or to be precise: phone 201 sends 200 ok, but does no

Re: [sipx-users] Problem with calls via GW when modifying "DefaultCaller ID"

Really the info you provide is not enough. Can you provide xml traces suitable for sipviewer of good and bad calls? It would be a good starting point to trobleshoot a problem. http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Si pviewer Rgds, Nikolay. > -Original Messa

Re: [sipx-users] phone INVITE -> TRYING -> RINGING -> connect? but no audio/voice with PolyCom SoundPoint 650

I would guess that you do not have voice when calling 200->201 because there is no Ack sent by "200-phone". And in the trace there is one way RTP. I would say that you should have "one way audio", not "no audio". I don't use polycom phones and I can't advice if you use appropriate firmware versi

Re: [sipx-users] [sipx-dev] DTMF timeout option for Voicemail

I think, it would be usefull to have such option somewheere in UI. +1. Rgds, Nikolay. > FWD ed on user mailing list, would like to hear from > community about exposing voicemail settings in sipXconfig > (e.g. initial timeout, inter Digit Timeout) - kind of what we > have for autoattendant rig

Re: [sipx-users] Problem with calls via GW when modifying"DefaultCaller ID"

Have you had a look into the trace? Take a trace of the call (http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+S ipviewer), send it here, describe your problem in more details.. The following command may be helpful (as root): sipx-trace -a -o ~/test.xml `sipx-dialog-count

Re: [sipx-users] Problem with calls via GW when modifying "DefaultCaller ID"

I usually first describe problem here, then create an issue in the tracker (when I'm told to do it) Rgds, Nikolay. > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Niklas > Sent: Tuesday, February 01, 2011

Re: [sipx-users] 404 Not Found for Inbound Calls

Probably sipregistrar.log (at Debug loglevel) will be helpful. Can you send a zipped fragment containing your problematic call here? Rgds, Nikolay. > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of > Henry D

Re: [sipx-users] interconnect with CS1000

Done. http://track.sipfoundry.org/browse/XX-9387 > Should i create a jira for it? > > seems like a bug to me. y, i would create a jira ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-us

Re: [sipx-users] interconnect with CS1000

ip.nstel.ru:SipRedirectServer-13:B62EFB90:SipRegistrar:"[090-USERPARAM] SipRedirectorUserParam::lookUp 'sip:3892;phone-context=cdp@sip.nstel.ru' not in my domain - not modified" Is it a bug or is it expected behaviour? Should i create a jira for it? Thanks and regards

Re: [sipx-users] interconnect with CS1000

rg] On Behalf Of Nikolay Kondratyev Sent: Thursday, January 27, 2011 5:16 PM To: 'Discussion list for users of sipXecs software' Subject: Re: [sipx-users] interconnect with CS1000 The description of the "Strip User Parameters" check box in the registrar configuration is Remo

Re: [sipx-users] interconnect with CS1000

.1? In 4.2.0? in 4.4.0? Are there any users that use sipx with SC1000? Any ideas how to solve the problem would be greatly appreciated. Thanks and regards, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nikola

Re: [sipx-users] Audiocodes call handling

As far as I know, it is possible if you manage mp118 manually via web interface (not via sipxconfig plugin). Try to create a huntgroup on mp118. With "channel select mode", say, ascending. Then in the "endpoint phone number table" you should mark four of your channels to belong to this huntgroup. T

Re: [sipx-users] interconnect with CS1000

t: Re: [sipx-users] interconnect with CS1000 Did you look at: http://track.sipfoundry.org/browse/XX-7695 <http://track.sipfoundry.org/browse/XX-7695> On Thu, Jan 27, 2011 at 7:23 AM, Nikolay Kondratyev wrote: Hi all, i have a problem interconnecting sipx (4.2.1 and 4.4.0) to cs10

[sipx-users] interconnect with CS1000

Hi all, i have a problem interconnecting sipx (4.2.1 and 4.4.0) to cs1000 (version 7.00.20). The problem is that CS1000 sends invite in the following form: INVITE sip:4002;phone-context=cdp.udp@"mydomain":5080;maddr=;transport=tcp;user=phone;x-nt-redirect=redirect-server SIP/2.0 and finally sip

[sipx-users] 4.4.0: sending fax?

Hi all, during testing fax feature in 4.4.0 i successfully received a fax. I.e. sipx user got an email with a tiff file. The fax was originated from mp114. T38 was used. Also i wonder if sipx user can send a fax message? via email-to-fax may be... or upload a file to a user portal and sending f

[sipx-users] 4.4.0 - sipxproxy failed

Hi, all. I started to play with 4.4.0. Today I upgraded my test system from 4.2.1 to 4.4.0 (4.4.0- 2011-01-18EST13:19:19 swift). And now sipxproxy process fails (sipxproc shows "failed" for sipxproxy). I tried to analize the problem: I found the following in the sipXproxy log: "2011-01-20T12:42:

Re: [sipx-users] Outbound back in via mediant

I would say, that first you need to find out where the call failed? Are you sure the Invite was sent to mediant? Or did the call failed inside the sipx? (in this case it could just be a permission confguration issue). It's not clear enough at which point did you collected the wireshark trace... S

[sipx-users] sipxconfig spa942 plugin: 'cfwd busy dest' parameter default value.

Hi all, i found that by default sipxconfig plugin sets "Cfwd Busy Dest" incorrectly. By default the value is set to 'vm'. This leads to incorrect behaviour when DND mode is activated on spa942 phone. The thing is that, when DND is set, spa942 replies with "302 moved temporarily" with "Cfwd Busy

Re: [sipx-users] MWI analog phones on ATA

Audiocodes MP1xx devices can do it. Though i dont know if mwi support is included into sipxconfig plugin for MP. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Thursday, January 13, 2011 6:15 PM To: m...@gro

Re: [sipx-users] Force VMs to email

It's on the "Unified Messaging" page of a user configuraion... > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of > m...@grounded.net > Sent: Friday, January 14, 2011 2:21 AM > To: sipx-users > Subject: [sipx

Re: [sipx-users] external-address

May be it's worth to think about using FS on a separate server with 2 nic's as SBC. Regards, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Irena Dolovcak Sent: Wednesday, January 12, 2011 3:41 PM To: Discussion list

Re: [sipx-users] Suspicious calls

Somebody (something :) ) from your local lan might just sent invites directly to AC. Do you collect syslog or cdr from your AC? (that would be for certain). "IP to tel calls count" on the AC may be interesting... Regards, Nikolay. > -Original Message- > From: sipx-users-boun...@list.sipfou

[sipx-users] duplicated challenge

Hi all, i encountered "duplicated challenge" from sipx. I have a separate FS profile on the same machine as sipx, listening on 15080, which works as sip-h.323 gw between sipx and avaya ipoffice. I also have an ITSP (sipnet) connection through sipxbridge. When i call ipoffice -> sip-h.323 gw

Re: [sipx-users] Call forwarding at phone

Henry, Many gateways can handle proxy auth. I tested it with audiocodes. Just configure a dummy user on the sipx, give him desired permissions. Make your gateway to use this dummy user credentials for proxy authentication. Rgds, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org

Re: [sipx-users] hairpin transfer from/to pstn causes error :SipUtilities.java:837

Recently i have analogous (Internal server Error at sipxbridge with similar exeption in the log) problem. In my case the problem occurs (as far as i can localize it) when _sip._udp srv record points to different ip address then the a-record for the ITSP domain. So... just in case... i would check

Re: [sipx-users] Remote Workers

second c line? If so, would > this have, in turn, send rtp traffic to remote worker's wan > ip address? > > Thanks in advance > > On Thu, Dec 16, 2010 at 3:47 AM, Nikolay Kondratyev > wrote: > > That is the problem is just like I described - m3 uses the first &g

Re: [sipx-users] authorization theoretical question

> [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nikolay > > Kondratyev [k...@nstel.ru] > > > > I know that there are many sip gurus in the list... > > I encountered the registrar, that uses status 407 in reply > to Register instead of 401. > > What i manag

Re: [sipx-users] Remote Workers

Local IP] t=0 0 a=sendrecv > m=audio 30484 RTP/AVP 0 8 101 c=IN IP4 [SIPX Server WAN IP] > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > a=x-sipx-ntap:X[SIPX Server Local IP]-[SIPX Server WAN IP];186 > >

[sipx-users] authorization theoretical question

Hi all, I know that there are many sip gurus in the list... I encountered the registrar, that uses status 407 in reply to Register instead of 401. What i managed to find out in rfc 3261, is that (if i'm not mistaken) registrar should use 401. But what could be the consequences of this misbehaviou

[sipx-users] sipxbridge and srv records

Hi all, i heve made some tests (sipx 4.2.1) with an ITSP, that have DNS A record pointing to their website, and SRV records, pointing to their SBC (Acme), whose ip address is different from website. And i found that sipxbridge sends Register to correct address - address given by SRV record. Bu

Re: [sipx-users] Remote Workers

to see all network activity coming and > out of snom m3. If so, I am not sure how to do a wireshark > trace on that. > > Actually, I could get a sip trace from snom m3. Would that do? > > Thanks > > On Fri, Dec 10, 2010 at 10:36 AM, Nikolay Kondratyev > wrote: > &g

Re: [sipx-users] Remote Workers

Can you capture a trace via wireshark on the port where m3 is connected? The thing is that sipx does "media relay" for remote workers, but sipx does it in a bit complicated way... Not all phones understand the way, sipx does "media relay". I suspect that snom does send rtp to remote worker but it

Re: [sipx-users] Remote Workers

Roman, You definitely don't provide enough information. I encountered analogous problem. Oh... No, symptoms were analogous. It's not possible to say something without a trace Could you provide a trace? (I mean xml trace with this call only, search wiki for "sipviewer", if you don't know how t

Re: [sipx-users] dtmf via sip info

g efforts for the sake of compatibility with them. Rgds, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nikolay Kondratyev Sent: Wednesday, December 01, 2010 5:47 PM To: 'Discussion list for users of sipXecs software'

Re: [sipx-users] session expiry

Will this patch be available in 4.2.1 (or may be 4.2.2) binaries? (via yum update) Thanks and regards, Nikolay. > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of > Douglas Hubler > Sent: Thursday, December

Re: [sipx-users] dtmf via sip info

I forgot to say, that sipnet does accept 2833 dtmf from sipx. But they always send dtmf via Info. That is the only problem is to receive dtmf via Info. Rgds, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nikolay

[sipx-users] dtmf via sip info

Hi all, the most famous ITSP here in Russia - Sipnet - uses sip info for sending dtmf digits, when sipx understands only 2833. It is not possible to make them change their mind and use 2833 There are about 15 ip pbx vendors listed in sipnet compatibility list. I believe sipx should (even m

Re: [sipx-users] SipX 4.2 media relay

sipXecs software > Subject: Re: [sipx-users] SipX 4.2 media relay > > > The main reason is that we have a couple of phones at remote > locations connected via VPN. Since we are in a rural area > with slow ADSL1 as our only Internet connection option, > having the audio from

Re: [sipx-users] SipX 4.2 media relay

DSL1 as our only Internet connection option, having the audio from these offices come in through the VPN and then back out to the ITSP is less than ideal. With QOS on the links quality is okay but latency is terrible and our bandwidth is very limited. Regards, Andrew Radke On 01/12/2010, at 7:4

Re: [sipx-users] SipX 4.2 media relay

from our ITSP, but we do need to authenticate outgoing calls via them. Regards, Andrew Radke On 01/12/2010, at 7:00 PM, "Nikolay Kondratyev" wrote: You mean register to ITSP? No, it can't (afaik). Rgds, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org [mailto

Re: [sipx-users] SipX 4.2 media relay

bject: Re: [sipx-users] SipX 4.2 media relay Can an unmanaged gateway authenticate with an ITSP though? Sorry, I didn't even look in 4.2 and I've left work for the day now. Regards, Andrew Radke On 01/12/2010, at 4:37 PM, "Nikolay Kondratyev" wrote: Andrew, do you use

Re: [sipx-users] SipX 4.2 media relay

Andrew, do you use "sip trunk" to connect to ITSP? my understanding is that, when you use siptrunk (sipxbridge), you get the following: 1. Refer messages are converted to re-Invite. 2. media relay 3. shortened headers (not so many via's). And you can not make sipxbriddge not to do it. If you really

Re: [sipx-users] FW: session expiry

2010 at 5:38 AM, Kris Amy wrote: Attached is the trace. From:sipx-users-boun...@list.sipfoundry.org <mailto:from%3asipx-users-boun...@list.sipfoundry.org> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nikolay Kondratyev Sent: Tuesday, 30 November 2010 7:05 PM To: 'Disc

Re: [sipx-users] session expiry

From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nikolay Kondratyev Sent: Tuesday, 30 November 2010 6:44 PM To: 'Discussion list for users of sipXecs software' Subject: Re: [sipx-users] session expiry Kris, i'm

Re: [sipx-users] testing t38 fax with sipx.

m might do the trick. > I am going on vacation tomorrow, but I'll take a look at the > email thing once I get back. > (http://track.sipfoundry.org/browse/XX-9235) > > C > > On 11/17/2010 08:44 AM, Nikolay Kondratyev wrote: > > Yes. > > Rgds, > > Nik

Re: [sipx-users] Remote Worker DNS Configuration

for users of sipXecs software > Subject: Re: [sipx-users] Remote Worker DNS Configuration > > This sounds right. Is it a big deal to change domain name > now? Do I need to reinstall sipx? > > Thanks in advance > > On Wed, Nov 24, 2010 at 6:39 AM, Nikolay Kondratyev > wro

Re: [sipx-users] Record all system prompts

When you record a vm messgae fs is used anyway. You may try to reconfigure current fs profile to accept HD audio from HD phone or create your own independent recoding service using independent fs sip profile, dialplan, e.t.c. http://wiki.sipfoundry.org/display/sipXecs/Custom+FreeSWITCH+programm

Re: [sipx-users] Remote Worker DNS Configuration

You could just configure external ip address, or external name, or both, as sip domain alias in the sipx. Then your remote workes will be able to register to external name and make/receive calls. But... Some features will not work. MWI and caller id substitution will not work. Possibly something

Re: [sipx-users] testing t38 fax with sipx.

undry.org] On Behalf Of > Nikolay Kondratyev > Sent: Monday, November 15, 2010 7:36 PM > To: 'Discussion list for users of sipXecs software' > Subject: Re: [sipx-users] testing t38 fax with sipx. > > I tried several times without success. > Trace and sipxivr.l

Re: [sipx-users] Record all system prompts

You could just leave a voicemail. Then take a file via user portal and save it with appropriate name. Then replace system files with yours. Or bettter: create your own localization pack with just voice prompts. Then you'll be able to switch between system and your own voice prompts very easily.

Re: [sipx-users] Caller ID settings Ignored in 4.2.1

Your phones register to "sip domain alias" not to "sip domain". In this case Caller ID feature will not work. I also encoutered this restriciton some time ago, I was told that sipxproxy (it is sipxproxy, who changes the From header) only looks for a "@sipdomain" in the From header, and does not loo

Re: [sipx-users] Sipx high availability with Mediant

I tested Mediant 2000 with HA sipx couple of years ago. I found some problems, but on the sipx side. There is a special configuration parameter on AudioCodes. Something like (i dont remember exact name) "dns resolve type", it can be 'a-record', 'srv-record' or 'na-ptr'. I tried both with 'srv-reco

Re: [sipx-users] Eyebeam intergrate with SIPX

Raymond, when you say : "it can not call" it means nothing. Please create a siptrace (http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+S ipviewer) xml file zip and post it in the list. Trace is needed to understand what is happening. As far as I know, sipx does not suppo

Re: [sipx-users] testing t38 fax with sipx.

Yes. Rgds, Nikolay. > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of > Cristi Starasciuc > Sent: Tuesday, November 16, 2010 6:47 PM > To: sipx-users@list.sipfoundry.org > Subject: Re: [sipx-users] testin

Re: [sipx-users] testing t38 fax with sipx.

: Re: [sipx-users] testing t38 fax with sipx. On 11/16/10 9:01 AM, Nikolay Kondratyev wrote: Ok, my yesterday problem were because of wrong usage of my fax sw. What fax software are you using? -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > | SECNAP Netw

Re: [sipx-users] testing t38 fax with sipx.

utoload_configs/fax.conf.xml and give it another try? Regards, Cristi On 11/15/2010 11:52 AM, Nikolay Kondratyev wrote: Hi all, i tried to test t38 fax with sipx/fs, ... but alas... I installed "a snapshot version" of sipx: 0.0.4.3.2-62363c1 2010-11-07T04:02:55 build27 on a virtua

[sipx-users] testing t38 fax with sipx.

Hi all, i tried to test t38 fax with sipx/fs, ... but alas... I installed "a snapshot version" of sipx: 0.0.4.3.2-62363c1 2010-11-07T04:02:55 build27 on a virtual machine, configured fax extention 301 for a user 201. Registered Audiocodes mp114 fxs as another user - 203. mp114 is configured to swi

Re: [sipx-users] Problem with directed call pickup

I guess that what you see is normal. I would advise to take closer look at the trace of working call pickup. Sipx uses (rather complicated) signalling procedure that is quite different from asterisk case. So ... make a call from phone1 to phone2. Pickup a call with phone3. Examine signalling with

Re: [sipx-users] We released our free ClickToDial Tool for SipX

Ability to dial sip uri would also be usefull... Thanks and regards, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nikolay Kondratyev Sent: Thursday, November 11, 2010 2:25 PM To: 'Discussion list for use

Re: [sipx-users] when to reload Freeswitch dial plans

+1 for the idea. But I'd like to offer to think a bit different way: will it be helpful to use different/independent sofia profile, listening on different port, say 15160, and different/independent context in the FS dialplan for openACD integration? Just an idea... Rgds, Nikolay. > -Original

Re: [sipx-users] We released our free ClickToDial Tool for SipX

Rene, Great tool! Thanks! Looks to work ok. But i found an issue: It does not allow me to change hotkey. I just show the following : I use win xp sp3. And i also have two improvement requests right now: 1. Use DNS name for sipx server. Sip domain name (SRV) would be the best way, imho, or sipx s

Re: [sipx-users] External audio player

I just tried to listen to vm message using Chrome. It shows buildin player picture, but nothing is audible. Rgds, Nikolay. > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of > George Niculae > Sent: Thursday

Re: [sipx-users] ITSP Timed Out problem

at 5:55 AM, Nikolay Kondratyev wrote: I also had analogous problems whith sipxbridge in 4.0.4. After migration to 4.2.1 i do not see the problem. I think it's worth upgrading to 4.2.1. Rgds, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list

Re: [sipx-users] ITSP Timed Out problem

I also had analogous problems whith sipxbridge in 4.0.4. After migration to 4.2.1 i do not see the problem. I think it's worth upgrading to 4.2.1. Rgds, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher

Re: [sipx-users] Route by Caller-id ... freeswitch

Matt, as far as i understand, you can just use caller_id_number instead of destination_number. See http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition. You could create separate FS profile, on sipx route a call to it (using unmanaged g

Re: [sipx-users] 4.2.1 zultis phone registration problem

y.org] On Behalf Of > Douglas Hubler > Sent: Thursday, October 21, 2010 4:55 PM > To: Discussion list for users of sipXecs software > Subject: Re: [sipx-users] 4.2.1 zultis phone registration problem > > On Thu, Oct 21, 2010 at 7:36 AM, Nikolay Kondratyev > wrote: > >

[sipx-users] 4.2.1 zultis phone registration problem

Hi all, I found that Zultys ZIP2x2L - 1.2.28 phone does not work with 4.2.1. Sipx rejects to register it. Something wring with digest authorization Phone sends Registers, gets 401, sends Register with Authorization... and gets 401 again, sends Register with Authorization and so forth... It w

Re: [sipx-users] Help with SUBSCRIBE for MWI

In fact I don't know, what exactly does that mean. I'm not a developer. May be a snapshot, containing all the logs and configuration will help. Create a snapshot, don't forget to restrict the snapshot by time. Make the snapshot available for downloading. Provide callid of the problematic subscrip

Re: [sipx-users] Auto Attendant speak twice even if Replay Countset to"1"

> Nicolay, are there easy way to retain localisation for sound > files but switch web back to english? > > Thanks, > Alexander It's not a problem. It is your browser, who is responsible for choosing a language. For IE: go to tools->internet options->languages. Put "english" on the first place.

Re: [sipx-users] Auto Attendant speak twice even if Replay Count set to"1"

For me it looks logical. Play_count = 1 + REplay_count. Rgds, Nikolay. > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of > ? > Sent: Wednesday, October 20, 2010 12:03 PM > To: sipx-users >

Re: [sipx-users] delete VM storage directory of the user upon userdeletion

On the other hand... Administrator allready has the ability to store a voicemail of all users - regular backup. So i'm in doubts if there is a need to preserve voicemailbox of a user when a user is deleted Rgds, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-use

Re: [sipx-users] Help with SUBSCRIBE for MWI

ones are trying to subscribe to that, same result. A Snom phone is able to register just fine on that server. Am 19.10.2010 14:17, schrieb Nikolay Kondratyev: By the way, is "voip.ikt-bs.de" configured as sip domain in sipx, or domain alias? (Registration "to domain alias"

Re: [sipx-users] Help with SUBSCRIBE for MWI

ead. It could also be that it sends the credentials incorrectly in some way though it can register properly. Am 19.10.2010 10:06, schrieb Nikolay Kondratyev: David, i took a look at your trace. Openstage sends the following Authorization header Authorization: Digest use

Re: [sipx-users] Help with SUBSCRIBE for MWI

Jeff, At least message trace is needed to try to help you. Either post siptrace xml file (http://wiki.sipfoundry.org/display/xecsuserV4r2/Display+SIP+message+flow+us ing+Sipviewer) or tcpdump capture. In both cases filter a trace by callid. Rgds, Nikolay. > -Original Message- > From: sipx

Re: [sipx-users] Help with SUBSCRIBE for MWI

David, i took a look at your trace. Openstage sends the following Authorization header Authorization: Digest username="200",realm="voip.ikt-bs.de",nonce="6c3df3c358dec988cc2725fea8844db f4ca5b588",uri="sip:2...@voip.ikt-bs.de:5060;transport=udp",response="c11b411 1b4f8dfbb752983fbe3d9fe47",qop=a

Re: [sipx-users] trial call recording in sipx

ture SIP packets and really see what is going on)? Thanks, Mike On Thu, Oct 14, 2010 at 7:52 AM, Nikolay Kondratyev wrote: Hi all, acording to my investigations in the tracker, call recording feature is delayed for indefinite time. Call recording function is highly anticipated. Meanwhile F

Re: [sipx-users] trial call recording in sipx

recorded (other than if you were to capture SIP packets and really see what is going on)? Thanks, Mike On Thu, Oct 14, 2010 at 7:52 AM, Nikolay Kondratyev wrote: Hi all, acording to my investigations in the tracker, call recording feature is delayed for indefinite time. Call recording functi

Re: [sipx-users] merging sipx-trace files?

May be siptrace-merge will do what you want? Rgds, Nikolay. P.S. [r...@beaver ~]# head /usr/bin/siptrace-merge #! /usr/bin/perl # To run: # # siptrace-merge [selection-options] file.xml ... >merged.xml # # Takes several siptrace XML files and merges them into one. # Based on timestamps and br

[sipx-users] wiki page broken

Hi all, i found that wiki page http://wiki.sipfoundry.org/display/xecsuserV4r2/Software+SIP+-+H.323+gateway, that i wrote month ago somehow became corrupted. In many places parameter names disapeared. For example, instead of there is on the page. I'd like to correct that, but it looks like i

Re: [sipx-users] "Incoming calls destination"

Alexander, By the way, 4.3.x - is development release. All releases, where the second digit is odd are not stable. Latest stable is 4.2.1. If you are not going to test the very latest features - use 4.2.1. Even if you are going to test the very latest features, it may worth starting with the st

Re: [sipx-users] 4.2.1 - "Delete message" link doesn't work

I just tested this in my install. If i'm not logged in into my web ui, then "delete message" link really does not work. But when i loogged in, it works. Rgds, Nikolay. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew Kitc

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